I want to use nodejs to stream multiple audios which is auto playing in the background. Meanwhile, someone could connect to the stream and listen to it. The scenario is like radio station.
I guess i could use websocket to broadcast to the client, but how to pass the playing audio data or is there any module I can use.
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I don't know how to get started with this.
What I am trying to do is get a video + audio stream from front-end and host the live stream as mp4 thats accessible on browser.
I was able to find information on WebRTC, socket.io, rtmp, but I'm not really sure what tool to use / whats best suited for something like this?
also follow up question, my front-end is iOS app. So what format would I send the live stream to the server?
It depends on which live streaming protocol you want to play on the player, as #Brad said, HLS is the most common protocol for player.
Note: Besides of HLS, iOS native app is able to use fijkplayer or FFmpeg to play any format of live streaming, like HLS, RTMP or HTTP-FLV, even MKV. However, the most straight forward solution is HLS, only need a tag to play MP4 or HLS, and MSE is also a optional solution to use flv.js/hls.js to play live streaming on iOS/Android/PC, this post is about these protocols.
The stream flow is like this:
FFmpeg/OBS ---RTMP--->--+
+--> Media Server---> HLS/HTTP-FLV---> Player
Browser ----WebRTC--->--+
The protocol to push to media server, or receive in node server, depends on your encoder, by RTMP or H5(WebRTC):
For RTMP, you could use FFmpeg or OBS to push stream to your media server.
If want to push stream by H5, the only way is use WebRTC.
The media server coverts the protocol from publisher to player, which use different protocols in live streaming right now(at 2022.01), please read more from this post.
I want to build an internet radio station using nodejs. My architecture needs to be like the following, there is one producer who records live audio, this live audio data needs to be sent to the server. On the server-side, I need to save the live streams of audio data in some audio format(for future playback) and also simultaneously stream the live audio to multiple clients. Can somebody please point me towards some implementations or library available to achieve this? I have read several posts and stackoverflow answers but couldn't find anything related to my need.
Is webRTC needed? I don't want clients to get peer to peer connection as I also want to save the live audio on the server. Please any help would be appreciated.
I have an interesting problem, in short: how to share information between threads in NodeJS (12+).
The tech stack - in short also:
A remote/online streaming server, what producing an MP4 live stream
A client application what only consumes live view through RTSP over HTTP
A small NodeJS based application to get the MP4, transform it and pipe it back to the client.
.
The modules what I use:
NodeJS 12+
Request/fetch/https module
Express module
Stream module
The story:
I have an application, what has a gateway/relay role between two different system. One provide a live media stream (simple MP4(h264) stream) and another one supposed to consume it as RTSP over HTTP. The weird part is, the consumer client does not behave like any other player (like VLC or a webplayer), sometime - seemingly randomly - resend the request, sometime close the current request and resend it. So direct pipe not really working for this use-case.
I made a worker (from worker_threads), what hold a readable stream object, and when the client hit the request, I start populate the MP4 stream into the readable object in the worker, so even if the stream is does get a close or resend, it will not break the live media stream consuming process.
And wherever the client connect, I just would like to pipe the readable object for it.
Originally, I though a simple pipe from like request/fetch/http.get or FFMPEG would be enough, but the client could call the call between 3 seconds and 2 minutes.
.
So, my questions are, what could be the best solution, to pass back the data from the worker to the main and let reach the HTTP routing?
I had some idea like:
I know, I can have my own channel between the threads and can pass back-and-forth information, but waiting for message and keep up the process does block the app, as far as I know (worker.on('message', (stuff) => {});).
Using Socket.io to pass data back from the worker, populate the readable in the main, and pipe the readable at http level (fake shared object basically)
Creating a secondary http server what offer the media stream, then i will just relay this into the response (e.g.: gatewaying/proxying)
Looking up some proxy solution where I can just simply redirect and reshape thing, like the input mp4 transforms into RTSP stream and pipe it to the consumer response
Should I just "remember" to the active stream, and if its streamed by the remote server, always just using the same url, passing to FFMPEG and continue piping to the res?
Note:
I setted up all the headers to keep alive the connection, but seems the client software act as-is.
By default its using RTSP and RTP/TCP to consume video stream, but has option for RTSP over http.
Probably I overlook some trivial task for serving RTSP video from a remote live MP4, but I did not found any good example or source anywhere (everywhere the same 3 article re-shared basically)
I did not found any similar question, nor article anywhere (but checked out the nodejs ffmpeg play video at specific time and stream it to client).
I am developing a colaborative instrument playing game, where multiple users will play an instrument (a synthesizer or sample, using the WebAudio API). On my first prototype I've set up a keyboard that sends note/volume signals via Socket.io to the server, and when the server gets that signal it sends it back to all connected sockets, which will play the corresponding note.
You might have guessed it right: there's a massive amount of lag and inconsistency as to the order of arrival of notes.
What are some efficient ways that I can send the output of WebAudio to the server, and have it broadcast to all connected users, so I have some sort of consistency?
You could try using a MediaStream by adding an MediaStreamAudioDestinationNode to your audio node graph as a destination and use that stream with either WebRTC or RecordRTC to send to your server.
Here is some info I found you could look at.
It does talk about using the getUserMedia method, but both getUserMedia and MediaStreamAudioDestinationNode methods send out a MediaStream constructor. This info
has some ideas on how you could send a MediaStream to your sever. However it does say that it needs to be recorded first. Not when it's live and running.
Sending a MediaStream to host Server with WebRTC after it is captured by getUserMedia
I hope this helps :)
So I am trying to create a RTSP server that streams music.
I do not understand how the server plays a music and different requests get what ever is playing at that time.
so, to organize my questions:
1) how does the server play a music file?
2) how does the request to the server look like to get whats currently playing?
3) what does the response look like to get the music playing in the client that requested the music?
First: READ THIS (RTSP), and THEN READ THIS (SDP), and then READ THIS (RTP). Then you can ask more sensible questions.
It doesn't, server streams little parts of the audio data to the client, telling it when each part is to be played.
There is no such request. If you want, you can have URL for live streaming, and in RTSP DESCRIBE request, tell the client what is currently on.
Read the first (RTSP) document, all is there! Answer to your question is this:
RTSP/1.0 200 OK
CSeq: 3
Session: 123456
Range: npt=now-
RTP-Info: url=trackID=1;seq=987654
But to get the music playing you will have to do a lot more to initiate a streaming session.
You should first be clear about what is RTSP and RTP. The Real Time Streaming Protocol (RTSP) is a network control protocol designed for use in communications systems to control streaming media servers. where as Most RTSP servers use the Real-time Transport Protocol (RTP) for media stream delivery. RTP uses UDP to deliver the Packet Stream. try to Understanding these concepts.
then Have a look at this project.
http://sourceforge.net/projects/unvedu/
This a open source project developed by our university, which is used to stream video(MKV) and audio file over UDP.
You can also find a .Net Implementation of RTP and RTSP here # https://net7mma.codeplex.com/ which includes a RTSP Client and Server implementation and many other useful utilities e.g. implementations of many popular Digital Media Container Formats.
The solution has a modular design and better performance than ffmpeg or libav at the current time.